Fiter design is the process of designing a filter.

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EQ - Signal attenuation with IIR bandpass filter bank

I've been charged to make an audio equalizer in c++, and I've been learning DSP during the last months but I'm still a newbie, so the title may not explain exactly my problem... I was able to make it ...
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32 views

Filter implementation and alternatives

I'm learning of digital filters and found a question asking why a filter with a impulse response $$h[n]= \begin{cases} \frac{1}{n^2+3n+2} & n\geq 0 \\ 0 & n<0 \\ ...
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37 views

Possible to implement simple IIR filter with greater than 6 db/octave slope?

I am using the following IIR code as a low-pass filter in a software project, and it works great. However, I would like the slope of the filter to be much steeper as the one here seems to be very ...
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1answer
18 views

What is the Reference tap of an equalizer (FIR Filter)

Currently I'm using the built-in Equalizer (EQ) object from Matlab to equalize my signal. For the EQ there is a parameter called the Reference tap. With different value of the reference tap i got also ...
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25 views

Necessity of Cyclic Prefix in OFDM

I have a question on CP-OFDM. In recent years people are talking about filter bank approach where they are trying to filter out the out of band side lobes using filters. Besides they are presuming ...
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25 views

Any simple Digital All-Pass Filter Design Tools?

I would like to be able to implement an All-Pass filter with a frequency-dependent group delay. I need a maximum group delay at low frequency of about 20 samples (for Fs = 44.1kHz), and it needs to ...
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2answers
98 views

Which filter for an audio equalizer

I’m new to DSP and I have to do an audio equalizer in c++. I did a lot of research about it and tried some stuff in the last month, but I’m a little overwhelmed with all those informations and it’s ...
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26 views

Deriche recursive filter using OpenCV

Is it possible to create a Deriche recursive filter using opencv (the Deriche filter is a blur that minimizes noise and preserves edges for later edge detection)? I can create one using CPU code, but ...
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35 views

ECG noise filtering using FIR Bandpass Filter

I am new to signal processing and examine ECG data corrupted with noise. The clean data has a frequency spectrum like this. And the corrupted one looks like this (sorry for the different scale) ...
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1answer
32 views

Help with Hankel Tranform filters

I found a filter for Hankel Transform . As far as i understand this is effectively used as an approximation for J0(x) and J1(x) . But how will the change in input affect these filter terms (eg J0(r) ...
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1answer
32 views

Equiripple Filter-Design

I have a requirement to design minimax filter with linear programing (linprog in MATLAB). To build the filter I must choose vector $\omega$ (frequency sample), how ...
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1answer
56 views

Sampling frequency and oversampling factor

A quite naïve question. I need some clarification regarding choosing the sampling frequency and oversampling factor. The scenario is as follows: I have an OQPSK modulated sequence with symbol rate 2 ...
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21 views

Can't make a Wiener filter work

I have two wav signals, one desired and the other one an unwanted signal perturbing the first one. I'm trying to implement a Wiener filter to reduce the second one, but when using the sound function I ...
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2answers
50 views

Find Stopband Edge Frequency

I tried to find the stopband edge frequency.I know that the signal amplitude falls to -3 dB below the passband edge frequency.I examine a lot of DSP books so i don't any information about how to ...
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38 views

Half Sine Pulse Shaping design

The standard of 802.15.4 specfies the following 'The chip sequences representing each data symbol are modulated onto the carrier using O-QPSK with halfsine pulse shaping.' For pulse shaping the ...
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1answer
44 views

What windowed Sinc might be closest to a Parks–McClellan low-pass filter?

In another question on dsp.stackexchange, a statement was made that the output of MATLAB firpm() or firls() for an anti-alias low-pass filter was "close" to the coefficients of a windowed-Sinc ...
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35 views

time domain non stationary time series individual colored noise identification algorithm

One of the methods for identification of modulated colored (stationary) noises is that of lag 1 autocorrelation power law noise identification algorithm. This method can determine the dominant noise ...
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1answer
17 views

Calculating the network function of a filter

Given a system, that behaves as a 1st order filter with network function $H(s)$. We input: $$v_1 (t)=1+3\cos(10^4 t)$$ And we obtain as output: $$v_2(t)=1+1.5\cos \left(10^4 t ...
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1answer
35 views

Convert low pass continuous time filter design to bandpass, discrete time

I am trying to convert the continuous time transfer function of a second order lowpass Butterworth filter is given by: To a bandpass fourth order bandpass digital filter, I first apply the mapping ...
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20 views

Filtering with Gaussian Shape

I would like to know how to properly filter a cosinus wave signal with a Gaussian Filter. I thought to use 'conv' to perform it. My code as follows : ...
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1answer
39 views

Parks-McClellan - compiling C++ code from Iowa Hills page [closed]

I am trying to compile C++ code for FIR filter design using Parks-McClellan algorithm from http://www.iowahills.com/Example%20Code/NewParksMcClellan.txt . However, I cannot find the following header ...
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2answers
103 views

How do can we cancel out noises while listening voice on headphones?

It's a common problem that while talking with the another person on Smart phones with the help of regular headphones,we get very annoying sounds at background . The opposite person's voice gets mixed ...
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1answer
44 views

Help choosing coefficients for a simple FIR filter to extract DC component

I am a beginner in design of digital filters... I have just completed Coursera's Linear Circuits course! I have a signal which is sampled at 50 Hz where the DC component has a value of 800000 and ...
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1answer
43 views

What is sound output delay of a band pass filter at different freqencies?

I measured that a 2-pole band pass filter, given imput audio of different frequencies, outputs an audio signal with different time delay. At 440Hz, the time delay between of the input audio and the ...
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1answer
14 views

Concepts Surrounding First Two Samples in Difference Equation

I'm attempting to construct a difference equation in matlab for the function u(n) and find the auto-correlation matrix R. I'm following the technique as is defined by my text below: The system I'm ...
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2answers
56 views

Design Filter with Arbitrary Magnitude and Phase Response in Matlab

Does anyone know of a way to design a filter with arbitrary magnitude and phase response in Matlab?
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29 views

Phase Linearization

Does anyone know of some simple algorithms to linearize the phase response of a filter. I have a notch filter with a non-linear phase response. I basically need an all pass filter (so that the ...
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20 views

high amplitude filter for same frequency signals

I'm analysing quasi-periodic movement (push ups). Each push up can take 1-5 seconds with stops or jerky movement along the way. I already used Savitzky-Golay filter. I want to filter out low ...
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35 views

How many different filters are needed for in an iterated filter bank?

I'm trying to implement an iterated filter bank as described in the Subband coding chapter of the Really Friendly Guide to Wavelets. However, the text seems to suggest that The advantage of this ...
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1answer
47 views

Matlab: is a highpass filter what I want, to obtain my required spectrogram? How do I design/apply this? (pic included)

The axes are wrong in the spectrogram. I have an audio wav file, and I know the sampling rate. this is read into the audioData variable. The audio is 1 channel. selected window length I believe (can't ...
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1answer
73 views

FIR design using firpm (Parks-Mclellan)

Can you please help me with some ideas on how to implement an FIR filter in Matlab using Parks-Mclellan algorithm? My specifications are the passband cut-off frequency, the transition bandwidth. The ...
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29 views

Filter signal1 from evolving frequency response of signal2

I'm a composer (of music) starting out with signal processing. I'm also reasonably proficient in C at this point. I'm not even sure if what I'm trying to do is possible, because I can't yet find an ...
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1answer
145 views

FIR filter coefficients from frequency responses

I have frequency responses of FIR filter of order 13 from obtained 50 frequency samples. How can I get FIR filter coefficients of order 13? For example, in MATLAB a lot of possible ways I tried ; I ...
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21 views

Filtering an image with a helmholtz-type equation

I'm attempting to apply the following PDE as an image filter to smooth a discrete heightmap with a helmholtz-type equation as described in this paper. It seemed like an interesting alternative to a ...
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1answer
39 views

Why this two FIR filters have the same amplitude but different phase response

Why a filter of the form $$ 1+a_0 z^{-1} +a_1 z^{-2}+ a_2 z^{-3} $$ has the same amplitude response when in the reversed form $$ a_2+a_1 z^{-1} +a_0 z^{-2}+z^{-3} $$ but the phase response is ...
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1answer
72 views

Spectral leakage issue cannot solved using windowing

I am designing a frequency lowpass filter using FFT&IFFT. My test data is a time series with a large DC component (the pink curve). Before the FFT, the time series was detrended (curve fitting) to ...
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1answer
35 views

transfer function of wiener filter

I am trying to understand the function of a Wiener filter. I get that the Wiener filter minimizes the mean square error between the estimated random process and the desired process. Does this mean ...
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1answer
55 views

How to account for sample rate in a FIR Digital Filter?

As a learning exercise I am trying to simulate a simple notch filter with Matlab. For test purposes I use Matlab to simulate sinewaves of various frequencies and plot frequency vs. gain so I can see ...
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24 views

scaling issue when using scypy's firwin2 to get coefficients for a filter

I am trying to use SciPy's firwin2 command to get the coefficients that correspond to a filter of my creating. I took white noise in Audacity, applied an ...
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1answer
39 views

Tradeoff between Filter Order and Sections

I have designed a bandpass IIR butterworth filter with desired response using the MATLAB FDAtool with the configuration shown in the figure below Since this is multistage implementation, when I ...
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1answer
50 views

1D filter for speed input and noisy position sensor

I was tasked with designing a filter to smooth out a translating table. The system, we command a speed and a desired position, and measure it with two noisy position sensors. The commanded speed is ...
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65 views

Basic “filtering” in digital communications

I am completely stuck on solving some problems for my digital communications class. Junior level undergrad class, so fairly basic. I have been searching around all day for related articles etc but ...
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1answer
51 views

Matlab FIlter coefficients to CMSIS api

I am trying to implement iir Band Pass filter design in STM32F429i discovery kit. I konw that it needs coefficients to be generated from matlab and feed them into cmsis api to achivi filtered data. ...
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46 views

MATLAB band-stop filter design for room reverberation?

I took acoustic measurements of a room using sine swept method, and I also used Pioneer sc-37 to obtain more accurate data. I used RoomEQ wizard software and measured the impulse response of the ...
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1answer
35 views

How would I go about designing a difference equation for a given type of filter?

I have experience using design tools such as Xilinx's CoreGen and MatLab's DSP tool kit, but I have not really had much experience designing filters from the ground up. Reinventing the filter as it ...
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1answer
62 views

custom Nearly-Linear-Phase IIR filter computation gone wrong

I'm using the code in this book :Algorithms for the constrained design of digital filters with arbitrary phase and magnitude responses to compute nearly-linear-phase IIR EQs. fvtool() shows the ...
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1answer
110 views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
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1answer
45 views

Given Gain function, How to design a causual, stable and minimum phase IIR filter?

I am given $|H(\omega)|$, I wonder if minimum phase stable causal filter is unique and how to calculate it.
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1answer
37 views

How can slope of low-pass biquad be configured?

I have a biquad filter, configuread as a low-pass filter (using Zolzer's formulas, similar to these). It does not overshoot and decays nicely in what I assume is -40dB/dec. How can I device a ...
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1answer
59 views

Recursive Implementation of the Gaussian Filter

I'm trying to implement an IIR form to approximate the Gaussian Blur Filter. I'm working with the article "Recursive Implementation of the Gaussian Filter" by Ian T. Young and Lucas J. van Vliet. ...